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Download Witura GSM VOIP Gateway Series User manual
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GSM VOIP Gateway Series User Manual GSM VOIP Gateway Series 1. OVERVIEW 1.1 Introduction GSM VOIP Gateway Series Color VoIP Channel GSM Channel Gray 1 1 Gray 4 4 Hardware Parameter Remark Model WT-2208 Customized The GSM VOIP Gateway is a broadband relay gateway newly developed by Processor ARM9E 133Mhz VinTelecom Technology. It is a new product for seamless connection between DSP VPDSP101 95Mhz the GSM network and VoIP network. When the mobile phone SIM card is RAM 16M installed in the GSM VOIP Gateway, users can register the GSM phone to the FLASH 4M VoIP softswitch system. Through the GSM VOIP Gateway, users can realize the Power DC12V/3A±10% Input 100V to 240V uplink and downlink call s between the GSM network and the VoIP network. In GSM Band Default 900M/1800M Default Optional 850/1900M Customized addition, the GSM VOIP Gateway supports the transparent transmission of the 2 pieces call er number from the PSTN to the VoIP. The Power Consumption GSM VOIP Gateway features embedded SIP and H.323 protocols with flexible LED RUN, GSM, LAN, PC setting. The bi-directional password authentication (call authorization) and trust Network Adapter 2 100/10BASE-T list authentication greatly minimize the risk of charge losses and the flexible Weight 1.204Kg Without AC Adapter routing function can meet special requirements of various call forwarding. In Operating Temperature 0-40°C particular, the GSM VOIP Gateway supports multi device groups, with flexible Operating Humidity 40%-90% Not Congealed setting of large GSM gateway groups with different channel numbers. With its low price, excell ent voice quali ty, and powerful features, the GSM VOIP Gateway is the first choice for system integrators, traffic operators, and softswit ch manufacturers. Max. 20W Color Gray VoIP Channel 8 GSM Channel 8 5 Factory Default Parameter Table The GSM VOIP Gateway includes WT-2201, WT-2204 and WT-2208. Parameter Network LAN PC Default Value Dynamic IP (DHCP) Fixed IP 192.168.8.1 Password Admin user Admin 1234 Default Time Zone GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GMT+8 75 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series 1.2 Select Tool > Reset to restart the GSM VOIP Gateway. 4 Parameters Of Equipment Hardware Feature Parameter Parameter Remark Model Processor DSP RAM FLASH Power WT-2201 ARM9E 133Mhz VPDSP101 95Mhz 16M 4M DC12V/2A ±10% WT-2204 ARM9E 133Mhz VPDSP101 95Mhz 16M 4M DC12V/2A ±10% WT-2208 GSM Band Default 900M/1800M Optional 850/1900M Max. 5W RUN, GSM, LAN, PC 2 0.10KG 0-40°C Default 900M/1800M Optional 850M/1900M Max. 12W RUN, GSM, LAN, PC 2 0.45KG 0-40°C 40%-90% Not Congealed 40%-90% Not Congealed Power Consumption LED Network Adapter Weight Operating Temperature Operating Humidity 2pcs Protocols TCP/ IP V4 (IPV6 Automatic Adaptive) ITU-T H.323 V4 standard H.2250 V4 standard H.245 V7 standard H.235 standard (MD5, HMAC-SHA1) ITU-T G.711 Alaw/ULaw, G.729A, G.729AB, G.723.1 and GSM voice coding RFC1889 real-time digital transmission protocol Firewall penetration technology SIP V2.0 standard STUN Network management protocol (NMP) PPPoE PPP authentication protocol (PAP) Internet control message protocol (ICMP) TFTP agent protocol Hypertext transfer protocol (HTTP) Dynamic host configuration protocol (DHCP) Domain name system (DNS) User account authentication (via MD5) Out-band DTMF relay: RFC 2833 and SIP INFO 1.3 Hardware Feature ARM9E high-speed processor Input 100V to 240V Default Voice coding and voice digital signal processor Two 10/100MB Ethernet ports that support the IEEE 802.3 standard and Customized PC LED that dis plays the status of Ethernet ports Ethernet cable SIM card that supports the GSM 900M/1800M and GSM 850M/1900M bands 100/10BASE-T Without 74 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD connect the LAN and GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Note: During the upgrading, do not cut off the power. Otherwise, the GSM VOIP 1.4 Software Feature Gateway will be damaged. LINUX OS Embedded HTTP that accesses internal parameters PPPoE diali ng You can modify the password of the user and administrator. Select Tool > Modify NAT broadband routing function Password. The password modification page is displayed, as shown in the following DHCP cli ent figure. Enter a new password and click “Change”. Then, the password is DHCP server successfully modified. Software online upgrade Automatic call ing Supporting multiple languages Supporting outgoing SMS calls 1.5 3.13.2 Product Package List 1) A GSM VOIP Gateway 2) 12V/2A DC(For WT-2204) or 12V/3A DC (For WT-2208) transformer 3) An Ethernet cable (2 m) Note: The password modified by users wil l be cleared and restored to the factory default password after the factory settings are restored. 3.13.3 1.6 Modification of Password Restore Factory Settings Product Appearance Select Tool > Restore Factory Settings. The following prompt is displayed. Click OK. All the parameters of the gateway will be cleared and the gateway will automatically restart. After the gateway is restarted, all the settings restore to the factory default settings. This feature can be completed by using the asterisk command. For details about this operation, see the section of “Asterisk Instructions”. 3.13.4 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD Reset 73 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Note: Some of parameters of the gateway wil l not be valid until the gateway is restarted. Therefore, you are advised to restart the gateway after the parameters are modified, so that the modification can take effect. 3.12 Abandon the change When the new setting is not saved, you can clear all the unsaved parameters. 3.13 Tool Select Menu > Tool. The following page is displayed. 1) LAN - The network input port that is connected to the router, Modem, and switch 2) PC - The network output port that is connected to network sharing 3) DC4.5V/2ADC - The output terminal that connects the transformer 4) Reset - The reset switch for quick restart of the GSM VOIP Gateway equipment (less than 100 terminals) 3.13.1 Online Upgrading equipped with delivery Warning! Only experienced users and administrators can implement the online upgrading. 2 Installation 2.1 Installation Procedure Select Tool > Online Upgrading. The online upgrading page is displayed, as shown in the following figure. Enter the complete name and path of the upgrade package, such as http://118.142.51.162/update/GHS-4.01-12.pkg, and then click “Start”. The gateway begins to upgrade. After the GSM VOIP Gateway is successfully upgraded, the gateway will automatically restart. The GSM VOIP Gateway has 1/8 SIM card slots, an LAN port, and a PC port. The install ation procedure is as follows: 1. Open the bottom cover of the GSM VOIP Gateway and insert an SIM card of the local GSM network 2. Connect the LAN port with the upper-layer network equipment with the Ethernet cable 72 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series 3. GSM VOIP Gateway Series PC port supports network sharing, so connect the PC port to the computer or lower-layer switch (HUB or router) 4. 2.2 Connect the output terminal of the transformer with the power port. Connection Figure 3.11 Save the Change After setting is changed, click “Save” and the new setting will be valid. Otherwise, the new setting is invalid. GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 71 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series SIP Terminal Disable: It’s not all owed to transfer the PSTN caller number to the VoIP system; Enable: The CID is set as the SIP caller number. Use Remote Party ID: The GSM VOIP Gateway will add the PSTN caller number to the call request signaling of the VoIP system. The signaling is as follows (provided that the PSTN caller number is 13800000000): 2.3 LED Indicators Use CID as SIP caller number: The GSM VOIP Gateway directly initiates the call request to the VoIP system through the PSTN caller number and adds the relevant information to Remote Party ID option of the request signaling. The call request signaling is as follows: 70 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series To: <sip:[email protected]> Call-ID: [email protected] CSeq: 3MESSAGE Contact: <sip:[email protected]> max-forwards: 16 The description of LED indicators is as follows LED date: Tue, 18 Nov 2008 06:36:37 GMT Description Power indicator LAN indicator Power indicator This indicator is constantly ON after connected with power p-hint: usrloc applied This indicator is ON after connected with the network equipment This indicator is ON after connected with the PC indicator user-agent: SIPPER for 3CX Phone network equipment and blinks during data transmission. 1. The RUN indicator blinks once every 100ms Content-Type: text/plain Content-Length: 26 13682626800 Hello world during startup. RUN indicator 2. When GSM VOIP Gateway is connected with the server, the RUN indicator bli nks once per Note: The SMS forwarding mode of the GSM VOIP Gateway is only functional under the SIP protocol. second. 1. When the GSM module of the GSM VOIP 3.10 Transparent Transmission of PSTN Caller Numbers Gateway logs onto the local GSM2. 2. Channel indicator When this indicator blinks quickly, it indicates The GSM VOIP Gateway permit s the transparent transmission of PSTN call er that the GSM VOIP Gateway is trying to log numbers to the VoIP system in various methods. onto the local GSM network. 3. When the GSM channel of the GSM VOIP Gateway is activated, this indicator is normally ON. 2.4 H323 Terminal SMS Instructions GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 69 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series The following is an example that the GSM VOIP Gateway forwards the SMS to the Users can send instructions to the GSM VOIP Gateway through the SMS. SIP 3999. The red part is the content of the SMS. Function MESSAGE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.2.162:5060;branch=z9hG4bK1967685528 From: <sip:[email protected]>;tag=667435795 Obtain information from the LAN port Reset the GSM VOIP Gateway setting To: <sip:[email protected]> Call-ID: [email protected] CSeq: 4MESSAGE Contact: <sip:[email protected]:5060> Reboot the GSM VOIP Gateway 1) case insensitive. The keyword REBOOT REBOOT Password is case insensitive. When “info” or “INFO” is sent to the GSM VOIP Gateway through SMS, the When performing reset/reboot through the SMS, the password a. The keyword “reset” and “reboot” is case insensitive, but the password is case sensitive. b. When the reset instruction is sent, the GSM VOIP Gateway will automatically reboot. 8613682626865 To perform reset, the admin password of GSM VOIP Gateway user is “tengda”. Input “reset tengda” or “RESET tengda” in the SMS to reset. 075583185700 To perform reboot, the admin password of GSM VOIP Gateway user is 2. The SMS sent to the GSM VOIP Gateway from the SIP is forwarded to the “tengda”. Input “reboot tengda” or “REBOOT tengda” in the SMS to reboot. specified PSTN number. MESSAGE sip:[email protected]:5060 SIP/2.0 The keyword RESET is RESET Password The detail ed procedure is as follows: Content-Length: 28 the SMS. Case insensitive authentication is required. The password is identical with that of the admin. Content-Type: text/plain indicates the number to receive the SMS. The second line indicates the content of INFO or info phone. 2) User-Agent: H 13682626800. Where, in the content of the SIP message (in red), the first li ne Remark GSM VOIP Gateway will return immediately the LAN port info to the mobile Max-Forwards: 30 The following example is about the Hello world sent from SIP 3999 to Instructions (Short message content) 3. Page Settings Before setting the page, you need to have the IP address of the PC port of the gateway first. Connect the computer for setting the gateway to the PC port of the GSM VOIP Gateway. The GSM VOIP Gateway has a built-in page server that is used to accept or obtain the HTTP. You can set the related functions for the GSM VOIP Gateway through the Internet Explorer. From: <sip:[email protected]>;tag=5031 68 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series 3.1 GSM VOIP Gateway Series Page Setting Menu canMapAlias = FALSE You can access the setting page of the GSM VOIP Gateway through the IP callIdentifier = { address of the LAN port or PC port. The default factory settings are as follows: guid = 16 octets { A: The LAN port supports the DHCP (dynamic IP address). Users can dial the SIM card number of the gateway and if connected dial *00 to obtain the IP cb 40 a4 af 8e 9b 60 96 6b 5f a0 03 f2 ed 55 5b .@....`.k_....U[ address. } B: The default IP address and mask of the PC port are 192.168.8.1 and 255.255.255.0 respectively. } Enable the DHCP service of the PC port. gatekeeperIdentifier = "GnuGk" Connect the computer with the PC port of the gateway through the IP address of willSupplyUUIEs = FALSE the PC port, and set the IP of the computer to dynamic IP or fixed IP as 192.168.8.xxx and the default gateway as 192.168.8.1. } 3.9.3 SMS Forwarding The GSM VOIP Gateway supports the SMS forwarding through the SMS under SIP protocol. After users send the short message to the GSM VOIP Gateway through the SMS, the GSM VOIP Gateway will send the short message to the specified VoIP number automatically. Open the Internet Explorer and enter 192.168.8.1 or http://192.168.8.1 in the address bar, then, the login page is popped up for password input. Enter the login account (“admin” as default) in the User Name, and password (“admin” as default) in the Password field. As shown in the above figure, select SMS Mode > Forwarding and enter the VoIP number that is used to receive the SMS information. The VoIP will automatically forward all the SMS information from the GSM network to this VoIP number. Similarly, the GSM VOIP Gateway will automatically forward all the SMS information from the VoIP to the specified GSM mobile phone. 1. The GSM VOIP Gateway forwards the SMS from the GSM to a specified SIP number. 10 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 67 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series [0] = dialedDigits "9998675588228822" } srcInfo = 2 elements { [0] = dialedDigits "20001" [1] = h323-ID "20001" } srcCallSignalAddress = ipAddress { Click “OK” button and the gateway status page is displayed as default ip = 4 octets { c0 a8 02 ed .... } port = 2049 } bandWidth = 2048 WT-2201 Status Interface callReferenceValue = 7502 conferenceID = 16 octets { 7f f3 78 77 49 3f 4c c1 9a dc 6a 84 12 d8 30 8f ..xwI?L...j...0. } activeMC = FALSE answerCall = FALSE 66 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 11 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series activeMC = FALSE answerCall = FALSE canMapAlias = FALSE callIdentifier = { guid = 16 octets { cb 40 a4 af 8e 9b 60 96 6b 5f a0 03 f2 ed 55 5b .@....`.k_....U[ } WT-2204 Status Interface } gatekeeperIdentifier = "GnuGk" willSupplyUUIEs = FALSE } When the user enters an SMS dial prefix, such as 999, the above call request signaling is changed to: Send RAS Message: admissionRequest admissionRequest { requestSeqNum = 241 WT-2208 Status Interface The GSM VOIP Gateway adopts the tree structure. The menu is on the left and the setting parameters are on the right, as shown in the above figure. callType = pointToPoint NULL endpointIdentifier = "3705_endp" destinationInfo = 1 elements { 12 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 65 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series callType = pointToPoint NULL GSM VOIP Gateway Series You can also access the setting page of the GSM VOIP Gateway through the IP address 192.168.2.216 or 192.168.2.172 of the LAN port of the gateway. The endpointIdentifier = "3705_endp" login method is the same as that of the PC port, but you must first obtain the IP address of the LAN port. destinationInfo = 1 elements { [0] = dialedDigits "8675588228822" 3.2 Status The status page contains the following contents, as shown in the above figure } srcInfo = 2 elements { [0] = dialedDigits "20001" 3.2.1 Telephone Information A. Product Sequence Number Each GSM VOIP Gateway [1] = h323-ID "20001" has a factory set sequence number, such as GOIP08030031, which is used for centralized setting, technical support, and maintenance filing. The sequence number is printed on the bottom plate of the } gateway and is read-only. srcCallSgnalAddress = ipAddress { ip = 4 octets { B. Software Version It displays the current version of software used by the GSM VOIP Gateway. When c0 a8 02 ed .... you want to upgrade the software, make sure the update version is newer than the current version. } port = 2049 C. Hardware Version It displays the current hardware version of the gateway. } D. Line Register Status bandWidth = 2048 It displays the login status of the line. When the line has logged into the SIP server callReferenceValue = 7502 or H.323 Gatekeeper, LOGIN is displayed, otherwise, LOGOUT is displayed. conferenceID = 16 octets { E. 7f f3 78 77 49 3f 4c c1 9a dc 6a 84 12 d8 30 8f ..xwI?L...j...0. It displays the use status of the line. When the line is in use, the status is ACTIVE, Line Use Status when the line is idle, the status is IDLE. } 64 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 3.2.2 Network Information 13 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series A. Examples of SMS Dialing: LAN Port In the following SMS dialing examples, the H.323 number of the GSM VOIP It displays the current IP address of the LAN port, such as 192.168.2.172. B. Gateway is set as follows: PC Port It displays the current IP address of the PC port. C. PPPoE Dialing It displays the PPPoE broadband connection condit ion. After the connection, the IP address obtained is displayed on the LAN port. D. Default Route Mode 2: It displays the current gateway address. E. Domain Name Server (DNS) It displays the current DNS address. 3.3 User Options Click “User Options” and the following figure shows. When a short message of “8675588228822” is sent from the mobile phone number (+86)13800000000 to the GSM VOIP Gateway, the GSM VOIP Gateway will send the following call requests. When the GSM VOIP Gateway sends a call request through the H.323 number of the GSM VOIP Gateway, the GSM VOIP Gateway will automatically add the number of the short message sender to the PSTN Forwarding Number in Call Forwarding (VoIP Incoming Call , Forwarding to the PSTN Immediately); In this mode, when the GSM VOIP Gateway receives the call from the H.323 GK, the GSM VOIP Gateway will forward the call to the short message sending equipment through the GSM network. The call request signaling in this mode is as follows: User Options of the WT-2201 Send RAS Message: admissionRequest admissionRequest { requestSeqNum = 241 14 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 63 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Content-Length: 226 3.9.2 SMS Dialing under the H.323 Protocol The GSM VOIP Gateway permits users to dial back through the SMS under the H.323 protocol. After users send the called number to the GSM VOIP Gateway through the SMS, the GSM VOIP Gateway will send a call request to the H.323 GK automatically. Users who need this function shall choose the following parameters: User Options of the WT-2204/ WT-2208 3.3.1 Language To select a language, refresh the page to enter the language page required. For Select SMS Mode > Dial and the following page is displayed. example, the current language is simplified Chinese. If you wish to display the page in English, click “English” in the menu. After your terminal is restarted, all The GSM VOIP Gateway supports three types of outgoing call via SMS: the pages will be displayed in Engli sh. A: Mode 1 (The current version of the H.323 protocol does not support this mode, but the later version will. ) In this mode, the GSM VOIP Gateway sets the number of the short message sender as the calling number of the call and the called number as the short message content; B: Mode 2 In this mode, the GSM VOIP Gateway sets the H.323 number of the GSM VOIP You can also use other languages for setting, as shown in the following figure Gateway as the calling number of the call and the called number as the short message content; C: Mode 3 (The current version of the H.323 protocol does not support this mode, but the later version will. ) In this mode, the GSM VOIP Gateway sets the H.323 number of the GSM VOIP Gateway as the calling number of the call and the called number as the short Click “English” on the upper right corner of the setting page. Then, the setting message content and the number of the short message sender, whose format is page will display all info in English. But the shortcut will not change the language short message content*the number of the short message sender. of the setting page when logged in next time. D: SMS Dial Prefix When the GSM VOIP Gateway initiates the SMS call, the GSM VOIP Gateway will change the prefix number to the called number prefix. 62 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 15 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series 3.3.2 Time Zone and Time Server GSM VOIP Gateway Series GSM VOIP Gateway sends a call request through the SIP number of the GSM VOIP Gateway, the GSM VOIP Gateway will automatically add the number of the short message sender to the PSTN Forwarding Number in Cal Forwarding (VoIP This item displays the adjusted time according to the selected time zone. The Incoming Call , Forwarding to the PSTN Immediately); gateway receive time and date information from the server through the Network Time Protocol and the time difference will l be automatically adjusted. For In this mode, when the GSM VOIP Gateway receives the call from the SIP server, example, the pacific standard time (PST) is GMT-8 and the pacific daylight time the GSM VOIP Gateway will forward the call to the short message sending (PDT) is GMT-7. equipment through the GSM network. The SMS dial prefix is still valid in this mode; The call request signaling in this mode is as follows: The time zone indicates the zone where the gateway is used. You need to enter the correct time zone, so that the time of the caller ID and charging information can be displayed correctly. The time server is the address of the server that obtains the network time through the Internet. The default time server is timekeeper.isi.edu. 3.3.3 DTMF Minimum Detection Interval This parameter is used to set the minimum interval of two DTMF signals. Packets may be lost during the data transmission over the GSM. As a result, a DTMF may be incorrectly identified as two or multiple identical DTMFs when detected by the GSM VOIP Gateway. The problem of repeated code can be solved effectively through the modification of the parameter. SendingMessage to 192.168.2.1:5060: INVITE sip:8675588228822*[email protected]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813 From: <sip:[email protected]:5060>;user=phone;tag=65248630 To: <sip:8675588228822*[email protected]> Call-ID: [email protected] CSeq: 2 INVITE Contact: <sip:[email protected]:5060> This parameter value ranges from 60ms to 120ms and the default value is 80ms. When the value of this parameter is increased properly, the repeated DTMF can Max-Forwards: 30 be avoided efficiently. However, the packet loss may also be caused. User-Agent:H 3.3.4 Automatic Setting Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, If the service provider provides the automatic setting, you can select “Enable” to MESSAGE, INFO, SUBSCRIBE start the automatic setting feature and enter the address of the server. If the service provider does provide the automatic setting, you need to select “Disable” Content-Type: application/sdp to speed up the startup time of the GSM VOIP Gateway 16 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 61 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series SendingMessage to 192.168.2.1:5060: INVITE sip:[email protected]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813 It is a special server, which needs the support of the specific system. From: <sip:[email protected]:5060>;user=phone;tag=65248630 To: <sip:[email protected]> Call-ID: [email protected] 3.3.5 Setting of the Remote Control Press *20# on the terminal to initiate the request to realize the remote management of equipment. The remote control server is provided by the service CSeq: 2 INVITE provider. The default port is 1920 and the terminal is identified by the SN. The remote control password is identical with that of the server and is set as default. Contact: <sip:[email protected]:5060> Max-Forwards: 30 User-Agent: H Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 226 In the following figure, the remote control server is set as 118.142.51.162. The terminal user presses *20# and a long tone is heard, which indicates that the instruction has been successfully sent. The remote administrator access http://118.142.51.162:8086 and the model and SN of the gateway are displayed. Click the gateway SN to set the remote gateway. Mode 3: When a short message of “8675588228822” is sent from the mobile phone Note: It is a special server, which needs the support of the specific system. For number (+86)13800000000 to the GSM VOIP Gateway, the GSM VOIP Gateway details, please contact the technical support Witura Corporation Sdn Bhd. wil l send the following call requests. When a short message of “8675588228822” is sent from the mobile phone number (+86)13800000000 to the GSM VOIP 3.3.6 GSM Group Mode Gateway, the GSM VOIP Gateway wil l send the following call requests. When the 60 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 17 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Users can establish a GSM group containing multiple GSM VOIP Gateway. Under this mode, the administrator only needs to provide a GSM number to the user to call in the VoIP system. From: <sip:[email protected]:5060>;user=phone;tag=65248630 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 2 INVITE Contact: <sip: [email protected]:5060> Max-Forwards: 30 User-Agent: H Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE Content-Type: application/sdp Each GSM VOIP Gateway can operate in any of the following modes: Content-Length: 226 Mode 2: Prohibit: This mode is used when the GSM VOIP Gateway operates independently. When a short message of “8675588228822” is sent from the mobile phone number (+86)13800000000 to the GSM VOIP Gateway, the GSM VOIP Gateway Serve as the server: When GSM VOIP Gateway operates in this mode, the will send the following call requests. When the GSM VOIP Gateway sends a call administrator only needs to provide the user with a GSM number of the GSM VOIP request through the SIP number of the GSM VOIP Gateway, the GSM VOIP Gateway as a unique access number to the GSM VOIP Gateway group. In one Gateway will automatically add the number of the short message sender to the GSM VOIP Gateway group, only one GSM VOIP Gateway can be used as the PSTN Forwarding Number in Call Forwarding (VoIP Incoming Call, Forwarding to server. When the GSM VOIP Gateway serves as the server, the GSM unconditional the PSTN Immediately); call forwarding or busy call forwarding can be activated. The unconditional call forwarding is used to forward all incoming calls to other clients of the group. The In this mode, when the GSM VOIP Gateway receives the call from the SIP server, busy call forwarding is used to forward incoming call s to other clients of the the GSM VOIP Gateway will forward the call to the short message sending group when the status of the SIM card of the server is ACTIVE. equipment through the GSM network. The SMS dial prefix is still valid in this mode; The call request signaling in this mode is as follows: 18 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 59 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Sending Message to 192.168.2.1:5060: INVITE sip:[email protected]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813 Serve as the client: When GSM VOIP Gateway operates in this mode, it will send it s real-time status to the server of the GSM VOIP Gateway group, so that From: the GSM VOIP Gateway server can deploy the call forwarding. <sip:[email protected]:5060>;user=phone;tag=65248630 Server address: It is the IP address of the GSM VOIP Gateway of the GSM VOIP Gateway group server. To: <sip:[email protected]> GSM number: It is the telephone number of the GSM SIM card used by the GSM Call-ID: [email protected] VOIP Gateway. 3.3.6 CSeq: 2 INVITE Anonymity Of the GSM Caller Number Contact: <sip: [email protected]:5060> Max-Forwards: 30 User-Agent: H Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 226 The caller number can be hidden, but this needs the support of the GSM operator. 3.3.7 IMEI When the user enters an SMS dial prefix, such as 999, the above call request signaling is changed to: The IMEI (International Mobile Equipment Identity) is an electric serial number containing 15 digits. 3.3.8 SMS Send to Client Sending Message to 192.168.2.1:5060: INVITE SIP/2.0 sip: [email protected]:5060;transport=udp Via: SIP/2.0/UDP 192.168.2.237:5060;branch=z9hG4bK363969813 58 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 19 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series C: Mode 3 In this mode, the GSM VOIP Gateway sets the SIP number of the GSM VOIP Gateway as the calling number of the call and the called number as the short message content and the number of the short message sender, whose format is short message content*the number of the short message sender. The SMS server sends the request to the mobile phone via the GSM VOIP Gateway, or the mobile phone sends the request to the SMS Server via GSM VOIP D: SMS Dial Prefix Gateway. When the GSM VOIP Gateway initiates the SMS call, the GSM VOIP Gateway will change the prefix number to the called number prefix. Examples of SMS Dialing: In the following SMS dialing examples, the account of the SIP of the GSM VOIP Gateway is set as follows: SMS server address: This is for filling the IP of the SMS server. Please make sure that server is installed with the GSM VOIP Gateway SMS management server software independently developed by Witura Corporation Sdn Bhd (The software can be obtained from our technical personnel). SMS server port: It is the SMS port of the SMS server. The default value is 44444. It must be consistent with that of the server if modified on the server. Authentication ID: It is the user ID. Make sure that the server has corresponding ID. Authentication password: It is for filling the user password. Note: The SMS send cli ent of the WT-2204 and WT-2208 are in the call set options, which needs the support of the GSM VOIP Gateway SMS management server. 3.3.9 GSM Band The GSM VOIP Gateway support the GSM/GPRS 900/1800 and 850/1900 bands. Mode 1: When a short message of “8675588228822” is sent from the mobile phone number (+86)13800000000 to the GSM VOIP Gateway, the GSM VOIP Gateway will send the following call request signaling: 20 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 57 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series 3.9 3.3.10 Timing Restart SMS Mode The GSM VOIP Gateway restarts at least once at the specified time every day to The GSM VOIP Gateway permits you to call VoIP users or forward short messages clear the buffer of the GSM VOIP Gateway, so that the GSM VOIP Gateway can through the SMS. operate normally. 3.9.1 SMS Dialing Under SIP Protocol Under the SIP protocol, the GSM VOIP Gateway permit s users to dial back through the SMS. After users send the called number to the GSM VOIP Gateway through the SMS, the GSM VOIP Gateway will send a call request to the SIP server automatically. Users who need this function should choose the following parameters: Select SMS Mode > Dial and the following page is displayed. The GSM VOIP Gateway supports three types of outgoing call via SMS: 3.3.11 China Area Code Matching The GSM VOIP Gateway can match all area codes of China to ensure the prompt dial. The default status is Disable. 3.3.12 IVR By default, the IVR is enabled. When a call comes in, the system prompts the user to dial a second time. When the IVR is disabled, the system will not prompt the user to dial a second time. 3.3.13 Prompt Tone System A: Mode 1 The prompt tones are the combination of the intervals and frequencies of the In this mode, the GSM VOIP Gateway sets the calling number of the SMS as the dialing tones and ring-back tones when users hook off the telephone. You can calling number of the call and the called number as the short message content; select the following prompt tones for the GSM VOIP Gateway according to the countries and regions where the GSM VOIP Gateway is used, so as to remain B: Mode 2 conventional tones. In this mode, the GSM VOIP Gateway sets the SIP number of the GSM VOIP Gateway as the calling number of the call and the called number as the short message content; 56 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 21 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series SIM card status reporting number: The gateway can report the status of SIM cards (remaining call duration) through the SMS. This parameter is used to specify the mobile phone number to receive the SMS. SIM card status reporting time: This parameter is used to specify the remaining call duration and then send the report. SIM card ID: This parameter is used to specify the ID of SIM cards in the short message report. You can specify the mobile phone number corresponding to the SIM card or any character string as the ID. Customize: One-time call duration limit of SIM cards: This parameter is used to specify the duration of one-time calls (by minutes). Users can customize prompt tones according to their special requirements. Select Customize, the following setting parameters are displayed. Examples and Explanations: The setting parameters are defined as follows: Each prompt tone involves the following parameters. If a parameter is not defined, the value of the parameter shall be 0. SIM Card Call Duration Setting SIM Card Remaining Call Duration <nc, rpt, c1on, c1off, c2on, c2off, c3on, c3off, f1, f2, f3, f4, p1, p2, p3, p4> The setting is same as the SIM card time limit diagram. The total call duration of nc: the number of tones (1-4) the SIM card is 30 minutes. When the call duration is less than or equals 10 minutes, the gateway will send a short message to 13713652130 (the SIM card rpt: the number of repeats (0 to infinity) ID is 2130 in the reporting message.) to report the remaining call duration. When one-time call duration is longer than 8 minutes, the call will be disconnected. c1on: the duration when the frequency 1 is on (ms) When the call duration is over, it becomes 0. Users can dial the SIM card number c1off: the duration when the frequency 1 is off (ms) by the mobile phone and when the second dialing tone is heard, press *10 to restore the value. c2on: the duration when the frequency 2 is on (ms) c2off: the duration when the frequency 2 is off (ms) 22 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 55 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series This mode is used to set the above password authentication and trust list authentication at the same time. c3on: the duration when the frequency 3 is on (ms) For a downlink call, the authentication mode is as follows: If the number in the C3off: the duration when the frequency 3 is off (ms) trust list is used to dial the user served by the PSTN, the call will be connected. When the number you dialed is not in the trust list, you need to enter the f1: the frequency of tone #1, 300 to 3000Hz password after the secondary dial tone is played. Then, the call will be connected. f2: the frequency of tone #2, 300 to 3000Hz For an uplink call, the authentication mode is as follows: If the mobile number or fixed number in the trust list is used to dial the user served by the VoIP, the call f3: the frequency of tone #3, 300 to 3000Hz will l be connected. When the number you dialed is not in the trust list, you need to enter the password after Please Enter the Password is played. Then, the call f4: the frequency of tone #4, 300 to 3000Hz will be connected. p1: the increment of tone #1, 0 to 31(0=3dB, -1dB increments) 3.8 Call Duration Limit p2: the increment of tone #2, 0 to 31(0=3dB, -1dB increments) The call duration limit is to limit the call duration of SIM cards in the gateway. Through this function, you can specify the total call duration of SIM cards. When p3: the increment of tone #3, 0 to 31(0=3dB, -1dB increments) the call duration is longer than the specified value, the call s shall not be connected to prevent the unnecessary or unsafe call charging. The setting p4: the increment of tone #4, 0 to 31(0=3dB, -1dB increments) parameters are as follows: Example: To add a prompt tone where f1 is 450Hz, c1on is 750ms, and c1off is 1000ms, enter the following values in the corresponding boxes: 1,0,750,1000,0,0,0,0,450,0,0,0,20,0,0,0 3.4 The parameters are defined as follows: SIM card limit time: This parameter sets the total call duration of SIM cards. Network Setting Click the “Network Setting” in the menu on the left, and the following page is displayed: When the call duration is longer than the specified time (by minutes), the call cannot be connected. When this parameter is null, the default call duration is infinite. 54 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 23 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Downlink 3.4.1 LAN Port Setting Uplink The setting is as follows: Select “Forward to PSTN Authentication Mode” > “Trust List Authentication”. Click the “VoIP Trust Number List”, and the VoIP Trust The LAN port of the GSM VOIP Gateway can be set to the dynamic IP through Number List is displayed (Maximum 15 trust numbers can be entered). Enter the DHCP, fixed IP, and PPPoE dialing. There are three setting modes: trust VoIP number in trust number sequence. If only a VoIP trust number, such as 3306, is set, only the number 3306 can be A. Dynamic IP (DHCP) used to dial the PSTN from the VoIP. This setting is default. If the network for the user provides the DHCP service, the GSM VOIP Gateway will require the network information such as IP address from 3.7.2.3 Password of Trust List Authentication the DHCP server automatically. B. Fixed IP Select fixed IP, and the following setting page is displayed. 24 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 53 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Set these parameters according to the network the user uses. 3.7.2 Authentication Mode Setting C. PPPOE The authentication mode is classified into the password authentication, trust list PPPoE (Point-to-point protocol over Ethernet) is a network protocol that authentication, and password or trust list authentication. compresses the PPP in the Ethernet. Select PPPoE dialing, and enter the account and password provided by the network provider. Downlink (VoIP to PSTN) authentication mode 3.7.2.1 Uplink (PSTN to VoIP) authentication mode Password Authentication D. 802.1qVLAN When the network serving the user provides the VLAN service, enter the parameter as required. The setting is as follows: Select “Forward to PSTN Authentication Mode” > “Password Authentication”. Enter the password in “Call PSTN Authentication E. Advance… Password”. Click Advance, and the Hardware address and Broadcast address are displayed in the page. As indicated in the above figure, for call s from the VoIP to the PSTN, when the second dialing tone is heard, dial the set password and the call will be connected. The hardware address is used to enter the MAC address in the format of XX: XX: For call s from the PSTN to the VoIP), when “Please Enter the Password” is played, XX: XX: XX: XX. enter the password and then the call will be connected. The broadcast address is used to communicate with other computers connected to the GSM VOIP Gateway. 3.7.2.2 Trust List Authentication 3.4.2 PC Port Setting The PC port can be set to connect other network equipment through the route or bridging mode. The two setting modes are as follows: A. Bridging When the PC port is set to the bridging mode, the relation between the LAN port and the PC port is layer 2 switching. The network equipment connected with the PC port same as the connection with the LAN port. 52 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 25 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series B. Fixed IP Address 3.7 Select the fixed IP, and the following setting parameters are dis played. Enter the Call Forwarding (Setting Authentication Mode) on the Call Route And IP address and subnet mask (the network section of the IP address should be different from that of the LAN port to prevent conflict). The gateway provides the call routing function for users, which can be set in the Call Forwarding Setting. The call routing is to forward calls to specified numbers, so that the dialing time can be decreased. In addition, the gateway provides three authentication modes for the uplink (call s from PSTN to VoIP) and downlink (call s from VoIP to PSTN). Do not set these parameters when they are not needed. 3.7.1 Call Route Setting C. Enable the DHCP Service This service can be enabled only when the PC port of the GSM VOIP Gateway is set as the fixed IP. To enable the DHCP service, you need to enter the start address and end address. D. Advance… Click Advance, and the Hardware address and Broadcast address are displayed in the page. XX: XX: XX: XX. The broadcast address is used to communicate with other computers that are connected to the ATA. Main DNS The DNS (domain name system) is a database that stores the Internet names and addresses, and converts between the name and the common Internet protocol digits. The main DNS is the IP address of the main DNS (such as 202.67.156.221 or obtain from the service provider). If the PPPoE is set, the main DNS will be automatically provided by the service provider. This parameter can be null . 3.4.4 Uplink: from PSTN to VoIP Note: The value of Call PSTN must be Enabling. Otherwise, The GSM VOIP Gateway prohibits any access to the PSTN. Therefore, set this parameter carefully. The above Note is also suitable for Call VoIP. 1. Set a hotline number in Call PSTN Forwarded-to-number. When the user The hardware address is used to enter the MAC address in the format of XX: XX: 3.4.3 Downlink: from VoIP to PSTN served by the VoIP network calls the GSM VOIP Gateway, the call is forwarded to the hotline number. When the user served by the VoIP network call s the GSM VOIP Gateway, the GSM VOIP Gateway connects the call and dials 88290211 directly. This feature is especially useful for hotline services. 2. Set a VoIP number in Call VoIP Forwarded-to-number. When the user served by the PSTN call s the VoIP network, the call is forwarded to the VoIP number. When a user calls another user served by the PSTN, the HT-342 calls the 3306 terminal of the VoIP. When the 3306 terminal answers the call , the HT-342 connects the call and the call will be connected. This feature enables international roamers to answer the phone through the VoIP anywhere. Secondary DNS 26 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 51 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series The rule is “0:|13[0-9]xxxxxxxx:+0|[1-8]xxxxxxx:+0755”. When the main DNS address fail s to connect or is not avail able, the secondary DNS can be used (such as 202.67.156.222 or obtain from the service provider). If When you dial the number 88990011 and 8899001133, the result is the same. the PPPoE is set, the secondary DNS will be automatically provided by the service The number actually dialed is 075588990011. provider. This parameter can be null . 3.6 Volume Adjustment 3.5 Call Setting This section describes the basic setting of the network connection relating to the The GSM VOIP Gateway has a built-in volume adjustment panel, which should be GSM VOIP Gateway, which supports two protocols: H323 and SIP. The setting used carefully. When you need to adjust the volume of the gateway, change the page is as follows: You can select a protocol in the “Terminal Type”. address http://xxx.xxx.xxx.xxx/xxx/gain.html to http://xxx.xxx.xxx.xxx/xxx/gain.html. Then, the following volume setting page is displayed. 3.5.1 H.232 Terminal Setting The H.323 protocol involves the direct connection mode and Gatekeeper mode. 3.5.1.1 Direct Connection After the volume setting is completed, click “Save” and the setting will take effect Under this mode, the GSM VOIP Gateway operates in the point-to-point status. immediately (for ongoing calls, the setting will not take effect immediately). The setting parameters are as follows: Note: The adjustment on the output volume of the line may cause the terminal to A. H.323Telephone Number fail in dialing numbers. Therefore, set this parameter carefully. The adjustment on The value of this parameter is a decimal numeral string that is used to confirm the input/output volume is for VoIP lines. the telephone number in the telephony network. For example, 5551234 is a valid telephone number. Enter the telephone number in this parameter. 50 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 27 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series B. Display Name This parameter is used to display the name of the user who subscribes the H.323 3.5.7.2 Dialing Rule with Specified Length Of Numbers service. For example, when you call your friend John Smith, your name will be displayed on your friend’s telephone. If you need to specify the length of telephone numbers matched, you can specify the dialing rule as “AAXXXXXX:-aa+bb”. Where, “AAXXXXXX” indicates the C. H.323 ID number to match and the length of the number. “AA” indicates the head and H232 ID is used to verify the account. Users can set this parameter according to other numbers are represented by X or x. The digit s after the colon indicates the requirements of the service provider. detailed actions to be taken on the number. D. Default Voice Gateway ay The setting is as follows: This parameter is used to find the proper Gatekeeper or IP address of the equipment of the call. Enter the IP address, such as 192.168.2.197, or domain In the above example 3, the rule is “00:|0:-0+0086|:+0086755”, which can be name, such as gk.yourisp.com. If the soft-switch system uses the non-standard changed to “00:|0:-0+0086|[1-8]xxxxxxx:+0086755”. port (1719 and 1720), you can add the detailed port number at the end of the IP address or domain name of the Gatekeeper. For example, if the port number is It means that when you dial a number whose first digit is 1 to 8 and total length 7300, is 8, the gateway will automatically dial the number and add 0086755 before the the IP address is 192.168.2.197:7300 and the domain name is gk.yourisp.com:7300. number. Under the direct connection mode, the GSM VOIP Gateway will forward all call s to Examples: the VoIP network to this address. "0:|13:+0|:+0755" Note: The value of this parameter must be standard ASCII characters (enter This rule enables the GSM VOIP Gateway to add 0 before the mobile phone characters under the English input mode). number and 0755 before the telephone number. 3.5.1.2 The above rule can be change to: Gatekeeper “0:|13[0-9]xxxxxxxx:+0|[1-8]xxxxxxx:+0755” Similarly, this rule enables the GSM VOIP Gateway to add 0 before the mobile phone number and 0755 before the telephone number. However, the length of the mobile phone number is limited to 11 digits. As shown above, the length of telephone numbers is limited to eight digits. 13[09]xxxxxxxx and [1-8]xxxxxxx represent mobile phone numbers 130xxxxxxxx to 139xxxxxxxx, and telephone number 1xxxxxxx to 8xxxxxxx, respectively. Note: When the length of the number is specified, the exceeded numbers will be discarded if the length of the number exceeds the specified length. For example: 28 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 49 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series added. If the number fails to match, the number continues to match the next Under the Gatekeeper mode, the GSM VOIP Gateway operates in the H.323 rule. If no digit after the colon is specified, such as “00:”, it indicates that no register status. When you register through the H.323 protocol, select “H.323 actions are taken when “00” is matched and the number exit s the matching. Terminal” in “Terminal Type”, as shown in the above figure. The registration If no digit before the colon is specified, such as “+86755”, it indicates that mode involves the Gatekeeper and direct connection (the direct connection mode instead of matching, actions are taken on the number directly. is used for call s over the IP address). When all lines use one number, select the 4. You can specify a range for matching of dialing rules. The rule format is “[A- single server setting mode. When all lines use different numbers, select the line B]A:-aa+bb” or “A[A-B]:-aa+bb”. For example, you can specify the range of setting mode. When you select the line setting mode, each line can be registered numbers beginning with 2 to 8 as “[2-8]: -aa+bb” or numbers beginning with to different servers. The setting parameters are as follows: 13 to 15 as “1[3-5]: -aa+bb”. A. H.323Telephone Number Examples: The value of this parameter is a decimal numeral string that is used to confirm 1. Rule: 0:|:+0755 the telephone number in the telephony network. For example, 191 is a valid a. The input number is “02083185711” and the output number is “02083185711”. telephone number. Enter the telephone number in this parameter. b. The input number is “83185700” and the output number is “075583185700”. B. Gateway ay Prefix 2. Rule: “00:-00|0"-0+86|:+86755” When you register through the gateway prefix, enter the prefix number. When a. The input number is “008522343318” and the output number is “8522343318”. b. The input number is “02083185711” and the output number is “862083185711”. c. The input number is “83185700” and the output number is “8675583185700”. the prefix number is called, a dialing tone is heard and then the secondary dialing is required. The gateway prefix enables the one-stage dialing. When users dial the gateway prefix and the telephone number, the gateway will automatically dial the number without the prefix. For example, the current gateway prefix is 123. If a user calls 075588290211, the user dials 123075588290211 on the IP phone. 3. Rule: “00:|0:-0+0086|:+0086755” a. The input number is “008522343318” and the output number “008522343318”. b. The input number is “02083185711” and the output number “00862083185711”. c. The input number is “83185700” and the output number “008675583185700”. is is is input number is “076322343318” and the output number is “13044557766” and the output number is “076322343318”. b. The input number is “013044557766”. Or, the input number is “13644557766” and the output number is “013644557766”. c. The input number is “23185700” and the output number is “075523185700”. Or, the input number is “73185700” This parameter is used to display the name of the user who subscribes the H.323 service. For example, when you call your friend John Smith, your name will be displayed on your friend’s telephone. D. H.323 ID 4. Rule: “0:|1[3-9]:+0|[2-8]:+0755|:+0755” a. The C. Display Name and the output number is “075573185700’. H232 ID is used to verify the account. You can enter this parameter according to the requirements of the service provider. E. Gatekeeper Address This parameter is used to find the proper Gatekeeper. Enter the IP address of the Gatekeeper, such as 192.168.2.197, or domain name, such as gk.yourisp.com. If the soft-switch system uses the non-standard port (1719 and 1720), you can add the detailed port number at the end of the IP address or domain name of the Gatekeeper. For example, if the port number is 7300, the IP address is 192.168.2.197:7300 and the domain name is gk.youris p.com:7300. 48 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 29 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series Under the Gatekeeper mode, the GSM VOIP Gateway will forward all calls to the C. STUN (RFC 3489) VoIP network to this address. The Simple Traversal of UDP over NAT (STUN) is a protocol that enables the SIP telephone to detect the existence and type of the firewall installed in the Note: The value of this parameter must be standard ASCII characters. computer. This parameter indicates the SIP address of the STUN server. Note: The STUN protocol supports the SIP gateway only. F. Enable VOS/AVS Encryption You can enable the VOS/AVS encryption. D. Trunk Agent The trunk agent protocol is a firewall traversal technology developed by Witura Corporation Sdn Bhd. It enables the products of by Witura Corporation Sdn Bhd to be applicable for most LANs. It involves the address, port, user name and password. The trunk agent protocol supports encryption on communications over the gateway. This feature needs the support of the server developed by Witura Corporation Sdn Bhd. Media agent mode: G. Enable Authentication on(Auth) Click “Enable Authentication” and enter the following parameters when you need Mode 1: media encryption and agent (supported by all versions of relay servers). to set the H.235 authentication code and password. Mode 2: media encryption and agent, supporting the transit over the specified port (supported by V2 relay server). Mode 3: media encryption and agent for the conversion of RTP data to TCP packet, supporting the transit over the specified port (supported by the relay server later thanV2). 3.5.1.3 Advance Setting of the H.232 3.5.7 Dialing Rules The advance option of the GSM VOIP Gateway involves the signaling and media, corresponding to “Advance Setting” and “Media” respectively. The GSM VOIP Gateway supports number dialing by rules. You can specify dialing rules in the dialing rule parameter of the “Call Forwarding”. Select “Advance Setting” under H.323 and the following setting page is displayed: 3.5.7.1 Format Of Dialing Rules 1. You can specify multiple rules, which are separated by the delimiter ‘|’. For example, “00:-00|0:-0+86|:+86755”. 2. The number is matched from the left of the dialing rules to the right. When the number matches the correct rule, the number stops matching. Otherwise, the number continues to match the next rule. 3. The rule format is “AA:-aa+bb”, such as “0:-0+86”. Where, “AA” indicates the number to match and “-aa+bb” indicates detailed actions to be taken on the number. If the number is successfully matched, “aa” is deducted and “bb” is 30 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 47 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series C. Trunk Agent The trunk agent protocol is a firewall traversal technology developed by Witura Corporation Sdn Bhd. It enables the products of Witura Corporation Sdn Bhd to be applicable for most LANs. It involves the address, port, user name and password. A. RAS Port The RAS is the communication protocol between the terminal and the Gatekeeper. It is used to transmit the registration information, login, broadband change, and The trunk agent protocol supports encryption on communications over the the status between two H.323. The RAS port can be used to specify the UDP and gateway. This feature needs the support of the server developed by Witura used with the router port for mapping. Corporation Sdn Bhd. B. Call Signaling Port (Q.931 Port) 3.5.6.3 Media NAT Traversal The media NAT (firewall) traversal is classified into four types: H.225-Q.931 is a call control protocol of the H.323 for transmits ting the call setting and unloading information between two H.323 units. It is used to specify the Q.931 port (TCP) that receives call s and used with the router port for mapping. A. No firewall The firewall traversal mechanism is not supported. C. Media Control ol Port (H.245 Port) H.245 is the media control protocol of the H.323. It is used to specify the port B. Port Transparency/DMZ that receives the H.245 connection (TCP) and used with the router port for The port transparency is used to transfer the network port on the LAN interface to mapping. the computer or the server in the LAN. This feature enables external users (through the Internet in most cases) to share the services of internal servers D. Fast Start (such as FTP, HTTP, and Telnet). You can enable or disable the fast start described in the H255.0 protocol. This parameter is used to detect and solve the compatibility problem. If you are not The port transparency supports the address of the gateway and response server. sure, do not set this parameter. The gateway is a communication device that connects two different networks. The response server is a standard service device that implements the ECHO protocol. E. Fast Start Extension This parameter is set for the special requirement of some customers. If you are not sure, do not set this parameter. 46 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 31 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series F. H245 Tunnel This parameter is set for the special requirements of some customers. If you are not sure, do not set this parameter. G. Registration Mode This parameter is used to comply with different PBXs and is not set normally. H. DTMF Signals DTMF signals are used to transmit call signals to the call switching center over the audio band. The DTMF means that two different frequencies of sounds are combined to 16 types of dialing tones. The telecom office or 1860 service hotline identify these dialing tones by analyzing the DSP, and thus determining the dialing number. There are two types of DTMFs: in-band DTMF and out-band DTMF. The trunk agent protocol supports encryption on communications over the gateway. The H323 trunk agent protocol supports encryption on signaling in different modes (for detail s about the agent mode, see section 3.5.6.3 “Media NAT Traversal”). Note: This feature needs the support of the server developed by Witura Corporation Sdn Bhd 3.5.6.2 Traversal Of SIP Signaling Over NAT The traversal of SIP signaling over NAT (firewall) is classified into: 1) In-band DTMF The in-band DTMF transmit s dialing tones and call voices together, without any processing. Therefore, the in-band DTMF transmits DTMF signals through a single way. 2) Out-band DTMF The out-band DTMF transmits dialing tones over protocols, such as RFC2833, which can ensure the validity of the transmission. A. No The mechanism of firewall traversal is not supported. B. STUN (RFC 3489) I. Signaling QoS Quality of Service (QoS) is a network’s capacity to provide priority services, including the special bandwidth, jitter control and delay (used for real-time and interactive traffic), and improvement of the packet loss ratio. This parameter is used to mark the specified QoS label for the call signaling packet to increase the network service quality. The Simple Traversal of UDP over NAT (STUN) is a protocol that enables the SIP phone to detect the existence and type of the firewall installed in the computer. This parameter indicates the SIP address of the STUN server. 32 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 45 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series In the advance option of the call setting, the signaling and media have separate firewall setting, as shown in the following figures. 3.5.6.1 Traversal of H323 Signaling Over NAT The traversal of H323 signaling over NAT (firewall) is classified into 4 categories: 3.5.2 SIP Phone The SIP (Session Initiation Protocol) is a simple network protocol that has less hierarchy and facilitates the initiation of calls among users. The call s may be conducted between two or more users, which include the sounds, images, session, A. No The mechanism of firewall traversal is not supported. B. Nat Citron The Citron is a special firewall traversal protocol for GnuGK and used with GnuGK. interactive games, and virtual reality. 3.5.2.1 Setting Mode The VoIP channel of the GSM VOIP Gateway can be set in the following three modes: single server, line setting, and trunk gateway. C. Port Transparency/DMZ The port transparency is used to transfer the network port on the LAN interface to the computer or the server in the LAN. This feature enables external users (through the Internet in most cases) to share the services of internal servers (such as FTP, HTTP, and Telnet). Figure 3-31 Setting Mode in the SIP Terminal A) Single server mode: Multiple VoIP channels can share the same setting. B) Line setting: Each VoIP channel can be served by different service providers The port transparency supports the address of the gateway and response server. The gateway is a communication device that connects two different networks. The response server is a standard service device that implements the ECHO protocol. D. Trunk Agent The trunk agent protocol is a firewall traversal technology developed by Witura or served by the same service provider. In the latter case, multiple different telephone numbers (accounts) can be registered on the same service, so that each telephone number is bound to the corresponding VoIP channel. C) Trunk Gateway: This mode is used to establish the connection or channel between the soft-switch and the gateway to realize the transit between two ends. Corporation Sdn Bhd. It enables the products of Witura Corporation Sdn Bhd to be applicable for most LANs. It involves the address of the trunk proxy server, port, user name and password. 44 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 33 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series 3.5.2.2 GSM VOIP Gateway Series Single Server Mode C. Jitter Delay Processing Mode This parameter is used to specify the algorithm model of the jitter delay buffer. The “Adaptive” mode should be set. Other modes are only used for tests and should not be set in actual applications. D. Media QoS Quality of Service (QoS) is a network’s capacity to provide priority services, including the special bandwidth, jitter control and delay (used for real-time and interactive traffic), and improvement of the packet loss ratio. This parameter is The setting parameters relating to the SIP are as follows: used to mark the specified QoS label for the voice packet to increase the network service quality. A) Telephone Number This parameter is used to set the telephone number of the line. The telephone number is a unique ID when the gateway serves as the caller. Note: For details about media encryption and media NAT penetration, refer to B) SIP Proxy Server This parameter is used to set the address of the SIP proxy server. If the SIP proxy server uses the special port (other than the SIP default port: 5060), you 3.5.5 Voice Coding And Sequence can add the detailed port number at the end of the IP address or domain name of This parameter is used to modify the compression coding according to the the proxy server. For example, 192.168.2.26:3000 or hy.con.com:3000. requirements of the service provider. C) SIP Registration Server The SIP registration server is a server used by the gateway to register the account. This parameter is used to set the IP address or domain name of the SIP login server. If the SIP registration server uses a special port (other than the SIP default port: 5060), you can add the detailed port number at the end of the IP address or domain name of the registration server. For example, 192.168.2.26:3000 or hy.con.com:3000. If a compression coding is ticked, it indicates that the compression coding is D) Outbound Proxy compression coding. available. The UP and DOWN are used to adjust the priority of the selected voice The outbound proxy is mainly used in the scenarios where the firewall or NAT exists, so that the signaling and media stream can penetrate the firewall . 34 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 3.5.6 Firewall Penetration 43 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series 2) Out-band DTMF E) Homing Domain The out-band DTMF transmit s dialing tones over protocols, such as RFC2833 and This parameter is used for the domain management host of the SIP (a host that SIP INFO, which can ensure the validity of the transmission. provides the SIP service). F. Registration Mode F) Authentication on ID When the registration information is sent to the platform under Mode 1, This parameter is used to set the authentication account when the gateway logs “experise” info is included. When the registration information is sent to the into the SIP proxy server. platform under Mode 2, “experise” variable is not sent. G) Password This parameter is used to set the authentication password when the gateway logs 3.5.4 Media Advance Setting into the SIP proxy server. H) Display Name The media advance setting is set for the RTP media stream of the gateway. Select When you call your friend John Smith, your name will be displayed on your “Call Setting” > “Media” and the following setting parameters are displayed: friend’s telephone. I) Backup Server This parameter is used for registration backup. When a backup registration server exists in the user’s system, the user can enable this parameter. Once the backup server is enabled, the gateway will automatically log into the backup server in case of the failure of the main server. A. RTP Port (Range) This parameter is used to specify the UDP of the RTP and used with the router port for mapping. Note: The terminal will use multiple pairs of RTPs (depending on the number of lines that the terminal supports). The value of this parameter ranges from 5500 to 5520. B. Packet Length (ms) This parameter indicates the duration of sending a network packet. If this parameter is null, it indicates that the default value is 20ms. It is used to specify the size of the media packet. The unit is ms (the actual number of bytes depends on the compression algorithm). 42 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 35 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series 3.5.2.3 Setting by Line (Valid for the WT-2204 and WT-2208) GSM VOIP Gateway Series C. Timeout Setting D. Signaling QoS Quality of Service (QoS) is a network’s capacity to provide priority services, including the special bandwidth, jitter control and delay (used for real-time and interactive traffic), and improvement of the packet loss ratio. This parameter is used to mark the specified QoS label for the call signaling packet to increase the network service quality. WT-2204 config by line E. DTMF Signals DTMF signals are used to transmit call signals to the call switching center over the audio band. The DTMF means that two different frequencies of sounds are combined into 16 types of dialing tones. The telecom office or 1860 service hotline identifies these dialing tones through analyzing the DSP, and thus determines the dialing number. There are two types of DTMF signals: in-band DTMF and out-band DTMF. WT-2208 config by line You need to set parameters for each line and the setting method is the same. The setting parameters are as follows: A) Telephone Number This parameter is used to set the telephone number of the line. The telephone 1) In-band DTMF The in-band DTMF transmit s dialing tones and call voices together, without any processing. Therefore, the in-band DTMF transmits DTMF signals through a single way. number is an unique ID when the gateway serves as the caller. 36 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 41 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series B) Gateway Prefix The gateway prefix enables the connection of call s through a particular line. It 3.5.3 SIP Advance Setting can match the first digit only. You can set a gateway prefix for multiple lines. When you set a gateway prefix for multiple lines, the call s that have the same The advance setting of the SIP involves the signaling and media. Users can set gateway prefix will select the line set with this gateway prefix. For example, the according to their special requirements. gateway prefix is 1. When the user dials 10086, the call will be connected by the line with the gateway prefix 1. When the user dials 075588290211, the system Select SIP Menu > Advance Setting/Media. detects whether the line with gateway prefix 0 exists. If exists, the call will be connected. Otherwise, the call will be released. Note: When you set the GSM VOIP Gateway by lines, the gateway prefix must be set. Otherwise, the call will not be connected. C) SIP Proxy Server This parameter is used to set the address of the SIP proxy server. If the SIP proxy server uses the special port (other than the SIP default port 5060), you can add the detailed port number at the end of the IP address or domain name of the proxy server. For example, 192.168.2.26:3000 or hy.con.com:3000. D) SIP Registration Server The SIP login server is a server that the gateway registers the account. This A. Signaling Port (SIP Local Port) The SIP local port is the local UDP port. It is used for communications between the SIP agent and the SIP proxy server as well as other SIP managers. B. NAT Hold This parameter is used to hold the port that is activated by the NAT for SIP signaling communication. The unit of the parameter is m. parameter is used to set the IP address or domain name of the SIP registration server. If the SIP registration server uses the special port (other than the SIP default port 5060), you can add the detailed port number at the end of the IP address or domain name of the registration server. For example, 192.168.2.26:3000 or hy.con.com:3000. E) Outbound Proxy The outbound proxy is mainly used in the scenarios where the firewall or NAT exists, so that the signaling and media stream can penetrate the firewall. F) Homing Domain This parameter is used for the domain management host of the SIP (a host that provides the SIP service). G) Authentication ID This parameter is used to set the authentication account when the gateway logs into the SIP proxy server. 40 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD 37 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD GSM VOIP Gateway Series GSM VOIP Gateway Series H) Password A) SIP Trunk Gateway1 This parameter is used to set the authentication password when the gateway logs It is the IP address of the server connected to the GSM VOIP Gateway. When the into the SIP proxy server. registration timeout is 0, the GSM VOIP Gateway is connected to the SIP server. If the registration timeout is not 0, the GSM VOIP Gateway logs into the SIP I) Display Name Trunk Gateway1 server through setting the telephone number, authentication ID When you call your friend John Smith, your name will be displayed on your and password. friend’s telephone. B) SIP Trunk Gateway2 J) Backup Server It is the IP address of the terminal connected to the GSM VOIP Gateway, which This parameter is used for registration backup. When a backup registration server can be an IP segment, such as 192.168.2.X. This means that all terminals exists in the user’s system, the user can enable this parameter. When the backup connected over 192.168.2 segment can log into the GSM VOIP Gateway and land server is enabled, the gateway will automatically log into the backup server in through the direct connection between the GSM VOIP Gateway and the GSM case of the failure of the main server. network. C) SIP Trunk Gateway3 It is the IP address of the server connected to the GSM VOIP Gateway. D) Telephone Number This parameter is used to set the telephone number of the line. The telephone 3.5.2.4 Trunk Gateway Mode number is an unique ID when the gateway serves as the callee and takes effect when the GSM VOIP Gateway logs into the SIP Trunk Gateway1. E) Registration Timeout (s) When the registration timeout is 0, you cannot register the gateway and the gateway will be connected to the server directly. You can reference the setting parameters of the single server mode to register the gateway. F) Authentication ID This parameter is used to set the authentication account when the gateway logs into the SIP Trunk Gateway1 proxy server. The parameter can be null in the case of interconnection. G) Password The trunk gateway is used to connect the VoIP network with the GSM network and convert the related protocols, so that users served by the two networks can call each other. 38 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD This parameter is used to set the authentication password when the gateway logs into the SIP Trunk Gateway1 proxy server. The parameter can be null in the case of interconnection. 39 GSM VOIP Gateway USER MANUAL – Rev1 – Technical Support: [email protected] COPYRIGHT ©2011 WITURA CORPORATION SDN BHD
